EP2070389B1 - Techniques facilitant le dialogue - Google Patents
Techniques facilitant le dialogue Download PDFInfo
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- EP2070389B1 EP2070389B1 EP07802317A EP07802317A EP2070389B1 EP 2070389 B1 EP2070389 B1 EP 2070389B1 EP 07802317 A EP07802317 A EP 07802317A EP 07802317 A EP07802317 A EP 07802317A EP 2070389 B1 EP2070389 B1 EP 2070389B1
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- H—ELECTRICITY
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- H04S3/00—Systems employing more than two channels, e.g. quadraphonic
- H04S3/008—Systems employing more than two channels, e.g. quadraphonic in which the audio signals are in digital form, i.e. employing more than two discrete digital channels
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- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
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- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
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- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
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Definitions
- Audio enhancement techniques are often used in home entertainment systems, stereos and other consumer electronic devices to enhance bass frequencies and to simulate various listening environments (e.g., concert halls).
- US Patent Publication No. 2005/117761 discloses a headphone apparatus allowing the listener to distinguish between two or more kinds of audio signals (e.g., video game audio and a player's voice) by supplying to a listener a first audio signal that is sensed as being located inside the head of the listener and a second sound signal subjected to signal processing such that the sound signal is sensed as being located outside of the listener's head.
- 2006 222686 discloses front and rear auxiliary speakers arranged opposite to a listener and performs "sound image localization control" for localizing sound images outputted from the auxiliary speakers at designated positions and "effect sound addition control” for adding effect sounds outputted from the auxiliary speakers to direct sounds outputted from main speakers.
- Other techniques attempt to make movie dialogue more transparent by adding more high frequencies, for example. None of these techniques, however, address enhancing dialogue relative to ambient and other component signals.
- a plural-channel audio signal (e.g., a stereo audio) is processed to modify a gain (e.g., a volume or loudness) of a speech component signal (e.g., dialogue spoken by actors in a movie) relative to an ambient component signal (e.g., reflected or reverberated sound) or other component signals.
- a gain e.g., a volume or loudness
- an ambient component signal e.g., reflected or reverberated sound
- the speech component signal is identified and modified.
- the speech component signal is identified by assuming that the speech source (e.g., the actor currently speaking) is in the center of a stereo sound image of the plural-channel audio signal and by considering the spectral content of the speech component signal.
- FIG. 1 is block diagram of a mixing model for dialogue enhancement techniques.
- FIG. 2 is a graph illustrating a decomposition of stereo signals using time-frequency tiles.
- FIG. 3A is a graph of a function for computing a gain as a function of a decomposition gain factor for dialogue that is centered in a sound image.
- FIG. 3B is a graph of a function for computing gain as a function of a decomposition gain factor for dialogue which is not centered.
- FIG. 4 is a block diagram of an example dialogue enhancement system.
- FIG. 5 is a flow diagram of an example dialogue enhancement process.
- FIG. 6 is a block diagram of a digital television system for implementing the features and processes described in reference to FIGS. 1-5 .
- FIG. 1 is block diagram of a mixing model 100 for dialogue enhancement techniques.
- a listener receives audio signals from left and right channels.
- An audio signal s corresponds to localized sound from a direction determined by a factor a .
- Independent audio signals n 1 and n 2 correspond to laterally reflected or reverberated sound, often referred to as ambient sound or ambience.
- Stereo signals can be recorded or mixed such that for a given audio source the source audio signal goes coherently into the left and right audio signal channels with specific directional cues (e.g., level difference, time difference), and the laterally reflected or reverberated independent signals n 1 and n 2 go into channels determining auditory event width and listener envelopment cues.
- the model 100 can be represented mathematically as a perceptually motivated decomposition of a stereo signal with one audio source capturing the localization of the audio source and ambience.
- FIG. 2 is a graph illustrating a decomposition of a stereo signal using time-frequency tiles.
- the signals S , N 1 , N 2 and decomposition gain factor A can be estimated independently.
- the subband and time indices i and k are ignored in the following description.
- the bandwidth of a subband can be chosen to be equal to one critical band.
- S , N 1 , N 2 , and A can be estimated approximately every t milliseconds (e.g., 20 ms) in each subband.
- STFT short time Fourier transform
- FFT fast Fourier transform
- the power of N 1 and N 2 is assumed to be the same, i.e., it is assumed that the amount of lateral independent sound is the same for left and right channels.
- the power ( P X1 , P X2 ) and the normalized cross-correlation can be determined.
- a , P S , P N can be computed as a function of the estimated P X1 , P X2 , and ⁇ .
- the least squares estimates of S , N 1 and N 2 are computed as a function of A , P s , and P N .
- a signal that is similar to the original stereo signal can be obtained by applying [2] at each time and for each subband and converting the subbands back to the time domain.
- g ( i , k ) is set to 0 dB at very low frequencies and above 8 kHz, to potentially modify the stereo signal as little as possible.
- FIG. 3A An example of a suitable function f is illustrated in FIG. 3A .
- the relation between f and A ( i,k ) is plotted using logarithmic (dB) scale, but A ( i , k ) and f are otherwise defined in linear scale.
- the constant W is related to the directional sensitivity of the dialogue gain.
- a value of W 6 dB, for example, gives good results for most signals. But it is noted that for different signals different W may be optimal.
- the function f can be shifted such that its center corresponds to the dialogue position.
- An example of a shifted function f is illustrated in FIG. 3B .
- the identification of dialogue component signals based on center-assumption (or generally position-assumption) and spectral range of speech is simple and works well in many cases.
- the dialogue identification can be modified and potentially improved.
- One possibility is to explore more features of speech, such as formants, harmonic structure, transients to detect dialogue component signals.
- a different shape of the gain function may be optimal.
- a signal adaptive gain function may be used.
- Dialogue gain control can also be implemented for home cinema systems with surround sound.
- One important aspect of dialogue gain control is to detect whether dialogue is in the center channel or not. One way of doing this is to detect if the center has sufficient signal energy such that it is likely that dialogue is in the center channel. If dialogue is in the center channel, then gain can be added to the center channel to control the dialogue volume. If dialogue is not in the center channel (e.g., if the surround system plays back stereo content), then a two-channel dialogue gain control can be applied as previously described in reference to FIGS. 1-3 .
- a plural-channel audio signal can include a speech component signal (e.g., a dialogue signal) and other component signals (e.g., reverberation).
- the other component signals can be modified (e.g., attenuated) based on a location of the speech component signal in a sound image of the plural-channel audio signal and the speech component signal can be left unchanged.
- FIG. 4 is a block diagram of an example dialogue enhancement system 400.
- the system 400 includes an analysis filterbank 402, a power estimator 404, a signal estimator 406, a post-scaling module 408, a signal synthesis module 410 and a synthesis filterbank 412. While the components 402-412 of system 400 are shown as a separate processes, the processes of two or more components can be combined into a single component.
- a plural-channel signal by the analysis filterbank 402 into subband signals i For each time k, a plural-channel signal by the analysis filterbank 402 into subband signals i.
- left and right channels x 1 ( n ), x 2 ( n ) of a stereo signal are decomposed by the analysis filterbank 402 into i subbands X 1 (i, k ), X 2 (i , k).
- the power estimator 404 generates power estimates of P ⁇ s , ⁇ , and P ⁇ N , which have been previously described in reference to FIGS. 1 and 2 .
- the signal estimator 406 generates the estimated signals ⁇ , N ⁇ 1 , and N ⁇ 2 from the power estimates.
- the post-scaling module 408 scales the signal estimates to provide ⁇ ', N ⁇ ' 1 , and N ⁇ ' 2 .
- the signal synthesis module 410 receives the post-scaled signal estimates and decomposition gain factor A, constant W and desired dialogue gain G d , and synthesizes left and right subband signal estimates ⁇ 1 ( i , k ) and ⁇ 2 (i,k) which are input to the synthesis filterbank 412 to provide left and right time domain signals ⁇ 1 (n) and ⁇ 2 ( n ) with modified dialogue gain based on G d .
- FIG. 5 is a flow diagram of an example dialogue enhancement process 500.
- the process 500 begins by decomposing a plural-channel audio signal into frequency subband signals (502).
- the decomposition can be performed by a filterbank using various known transforms, including but not limited to: polyphase filterbank, quadrature mirror filterbank (QMF), hybrid filterbank, discrete Fourier transform (DFT), and modified discrete cosine transform (MDCT).
- QMF quadrature mirror filterbank
- DFT discrete Fourier transform
- MDCT modified discrete cosine transform
- a first set of powers of two or more channels of the audio signal are estimated using the subband signals (504).
- a cross-correlation is determined using the first set of powers (506).
- a decomposition gain factor is estimated using the first set of powers and the cross-correlation (508). The decomposition gain factor provides a location cue for the dialogue source in the sound image.
- a second set of powers for a speech component signal and an ambience component signal are estimated using the first set of powers and the cross-correlation (510).
- Speech and ambience component signals are estimated using the second set of powers and the decomposition gain factor (512).
- the estimated speech and ambience component signals are post-scaled (514).
- Subband signals are synthesized with modified dialogue gain using the post-scaled estimated speech and ambience component signals and a desired dialogue gain (516).
- the desired dialogue gain can be set automatically or specified by a user.
- the synthesized subband signals are converted into a time domain audio signal with modified dialogue gain (512) using a synthesis filterbank, for example
- the dialogue boosting effect is compensated by normalizing using weights ⁇ 1 - ⁇ 6 with g norm .
- the normalization factor g norm can take the same value as the modified dialogue gain 10 g i k 20 .
- g norm can be modified.
- the normalization can be performed both in frequency domain and in time domain. When it is performed in frequency domain, the normalization can be performed for the frequency band where dialogue gain applies, for example, between 70 Hz and 8 KHz.
- input signals X 1 ( i , k ) and X 2 ( i , k ) are substantially similar, e.g., input is a mono-like signal, almost every portion of input might be regarded as S, and when a user provides a desired dialogue gain, the desired dialogue gain increases the volume of the signal. To prevent this, it is desirable to user a separate dialogue volume (SDV) technique to observe the characteristics of the input signals.
- SDV dialogue volume
- the normalized cross-correlation of stereo signals is calculated
- the normalized cross-correlation can be used as a metric for mono signal detection.
- phi in [4] exceeds a given threshold, the input signal can be regarded as a mono signal, and separate dialogue volume can be automatically turned off.
- the input signal can be regarded as a stereo signal, and separate dialogue volume can be automatically turned on.
- time smoothing techniques can be incorporated to get ⁇ (i,k).
- FIG. 6 is a block diagram of a an example digital television system 600 for implementing the features and processes described in reference to FIGS. 1-5 .
- Digital television is a telecommunication system for broadcasting and receiving moving pictures and sound by means of digital signals.
- DTV uses digital modulation data, which is digitally compressed and requires decoding by a specially designed television set, or a standard receiver with a set-top box, or a PC fitted with a television card.
- the system in FIG. 6 is a DTV system, the disclosed implementations for dialogue enhancement can also be applied to analog TV systems or any other systems capable of dialogue enhancement.
- the system 600 can include an interface 602, a demodulator 604, a decoder 606, and audio/visual output 608, a user input interface 610, one or more processors 612 (e.g., Intel® processors) and one or more computer readable mediums 614 (e.g., RAM, ROM, SDRAM, hard disk, optical disk, flash memory, SAN, etc.). Each of these components are coupled to one or more communication channels 616 (e.g., buses).
- the interface 602 includes various circuits for obtaining an audio signal or a combined audio/video signal.
- an interface can include antenna electronics, a tuner or mixer, a radio frequency (RF) amplifier, a local oscillator, an intermediate frequency (IF) amplifier, one or more filters, a demodulator, an audio amplifier, etc.
- RF radio frequency
- IF intermediate frequency
- filters filters
- demodulator an audio amplifier
- the tuner 602 can be a DTV tuner for receiving a digital televisions signal include video and audio content.
- the demodulator 604 extracts video and audio signals from the digital television signal. If the video and audio signals are encoded (e.g., MPEG encoded), the decoder 606 decodes those signals.
- the A/V output can be any device capable of display video and playing audio (e.g., TV display, computer monitor, LCD, speakers, audio systems).
- dialogue volume levels can be displayed to the user using a display device on a remote controller or an On Screen Display (OSD), for example.
- the dialogue volume level can be relative to the master volume level.
- One or more graphical objects can be used for displaying dialogue volume level, and dialogue volume level relative to master volume. For example, a first graphical object (e.g., a bar) can be displayed for indicating master volume and a second graphical object (e.g., a line) can be displayed with or composited on the first graphical object to indicate dialogue volume level.
- the user input interface can include circuitry (e.g., a wireless or infrared receiver) and/ or software for receiving and decoding infrared or wireless signals generated by a remote controller.
- a remote controller can include a separate dialogue volume control key or button, or a separate dialogue volume control select key for changing the state of a master volume control key or button, so that the master volume control can be used to control either the master volume or the separated dialogue volume.
- the dialogue volume or master volume key can change its visible appearance to indicate its function.
- the one or more processors can execute code stored in the computer-readable medium 614 to implement the features and operations 618, 620, 622, 624, 626, 628, 630 and 632, as described in reference to FIGS. 1-5 .
- the computer-readable medium further includes an operating system618, analysis/ synthesis fliterbanks 620, a power estimator 622, a signal estimator 624, a post-scaling module 626 and a signal synthesizer 628.
- the term "computer- readable medium” refers to any medium that participates in providing instructions to a processor 612 for execution, including without limitation, non- volatile media (e.g., optical or magnetic disks), volatile media (e.g., memory) and transmission media.
- Transmission media includes, without limitation, coaxial cables, copper wire and fiber optics. Transmission media can also take the form of acoustic, light or radio frequency waves.
- the operating system 618 can be multi-user, multiprocessing, multitasking, multithreading, real time, etc.
- the operating system 618 performs basic tasks, including but not limited to: recognizing input from the user input interface 610; keeping track and managing files and directories on computer-readable medium 614 (e.g., memory or a storage device); controlling peripheral devices; and managing traffic on the one or more communication channels 616.
- computer-readable medium 614 e.g., memory or a storage device
- the described features can be implemented advantageously in one or more computer programs that are executable on a programmable system including at least one programmable processor coupled to receive data and instructions from, and to transmit data and instructions to, a data storage system, at least one input device, and at least one output device.
- a computer program is a set of instructions that can be used, directly or indirectly, in a computer to perform a certain activity or bring about a certain result.
- a computer program can be written in any form of programming language (e.g., Objective-C, Java), including compiled or interpreted languages, and it can be deployed in any form, including as a stand-alone program or as a module, component, subroutine, or other unit suitable for use in a computing environment.
- Suitable processors for the execution of a program of instructions include, by way of example, both general and special purpose microprocessors, and the sole processor or one of multiple processors or cores, of any kind of computer.
- a processor will receive instructions and data from a read-only memory or a random access memory or both.
- the essential elements of a computer are a processor for executing instructions and one or more memories for storing instructions and data.
- a computer will also include, or be operatively coupled to communicate with, one or more mass storage devices for storing data files; such devices include magnetic disks, such as internal hard disks and removable disks; magneto-optical disks; and optical disks.
- Storage devices suitable for tangibly embodying computer program instructions and data include all forms of nonvolatile memory, including by way of example semiconductor memory devices, such as EPROM, EEPROM, and flash memory devices; magnetic disks such as internal hard disks and removable disks; magneto-optical disks; and CD-ROM and DVD-ROM disks.
- semiconductor memory devices such as EPROM, EEPROM, and flash memory devices
- magnetic disks such as internal hard disks and removable disks
- magneto-optical disks and CD-ROM and DVD-ROM disks.
- the processor and the memory can be supplemented by, or incorporated in, ASICs (application-specific integrated circuits).
- ASICs application-specific integrated circuits
- the features can be implemented on a computer having a display device such as a CRT (cathode ray tube) or LCD (liquid crystal display) monitor for displaying information to the user and a keyboard and a pointing device such as a mouse or a trackball by which the user can provide input to the computer.
- a display device such as a CRT (cathode ray tube) or LCD (liquid crystal display) monitor for displaying information to the user and a keyboard and a pointing device such as a mouse or a trackball by which the user can provide input to the computer.
- the features can be implemented in a computer system that includes a back-end component, such as a data server, or that includes a middleware component, such as an application server or an Internet server, or that includes a front-end component, such as a client computer having a graphical user interface or an Internet browser, or any combination of them.
- the components of the system can be connected by any form or medium of digital data communication such as a communication network. Examples of communication networks include, e.g., a LAN, a WAN, and the computers and networks forming the Internet.
- the computer system can include clients and servers.
- a client and server are generally remote from each other and typically interact through a network.
- the relationship of client and server arises by virtue of computer programs running on the respective computers and having a client-server relationship to each other.
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Claims (15)
- Procédé de traitement d'un signal audio, comprenant :l'obtention d'un signal audio à canaux multiples incluant un signal de composante de parole et un signal d'une autre composante ;la détermination de valeurs de gain pour au moins deux canaux du signal audio à canaux multiples, les valeurs de gain représentant un niveau pour chaque canal des au moins deux canaux ;la détermination d'une corrélation croisée entre les au moins deux canaux ;la détermination d'une localisation spatiale du signal de composante de parole en utilisant au moins l'une de la corrélation croisée et des valeurs de gain ;l'identification du signal de composante de parole sur la base de la localisation spatiale du signal de composante de parole ;la modification du signal de composante de parole en appliquant un gain au signal de composante de parole ; etla génération d'un signal audio modifié incluant le signal de composante de parole modifié.
- Procédé selon la revendication 1, où la modification du signal de composante de parole comprend en outre :l'identification du signal de composante de parole sur la base d'un domaine spectral du signal de composante de parole.
- Procédé selon la revendication 1, où le gain est une fonction de la localisation du signal de composante de parole et un gain désiré pour le signal de composante de parole.
- Procédé selon la revendication 3, où la fonction est une fonction adaptative de gain de signal ayant une région de gain qui se rapporte à une sensibilité directionnelle du facteur de gain.
- Procédé selon l'une quelconque des revendications 1, 2, 3 et 4, comprenant en outre :la normalisation du signal audio à canaux multiples avec un facteur de normalisation dans un domaine temporel ou un domaine fréquentiel.
- Procédé selon l'une quelconque des revendications 1, 2, 3, 4 et 5, comprenant en outre :la comparaison de la corrélation croisée avec une ou plusieurs valeurs de seuil ;la détermination si le signal audio à canaux multiples est sensiblement mono sur la base de résultats de la comparaison ; etla modification du signal de composante de parole lorsque le signal audio à canaux multiples n'est pas sensiblement mono.
- Procédé selon l'une quelconque des revendications 1, 2, 3, 4, 5 et 6, comprenant en outre :la décomposition (502) du signal audio à canaux multiples en un certain nombre de signaux en sous-bandes de fréquences, dans lequel:la détermination des valeurs de gain comprend l'estimation (504) d'un premier ensemble de puissances pour les au moins deux canaux en utilisant les signaux en sous-bandes,la détermination de la corrélation croisée comprend la détermination (506) de la corrélation croisée en utilisant le premier ensemble de puissances estimées, etla détermination de la localisation spatiale du signal de composante de parole comprend l'estimation (508) d'un facteur de gain de décomposition en utilisant le premier ensemble de puissances estimées et la corrélation croisée, dans lequel le facteur de gain de décomposition donne une marque de localisation du signal de composante de parole.
- Procédé selon la revendication 7, dans lequel la largeur de bande d'au moins une sous-bande est choisie de manière à être égale à une bande critique d'un système auditif humain.
- Procédé selon la revendication 7, comprenant en outre:l'estimation (510) d'un deuxième ensemble de puissances pour le signal de composante de parole et un signal de composante d'ambiance à partir du premier ensemble de puissances et de la corrélation croisée dans lequel l'autre signal de composante inclut le signal de composante d'ambiance.
- Procédé selon la revendication 9, comprenant en outre:l'estimation (512) du signal de composante de parole et du signal de composante d'ambiance en utilisant le deuxième ensemble de puissances et le facteur de gain de décomposition.
- Procédé selon la revendication 9, où les signaux de composantes de parole et d'ambiance estimés sont déterminés en utilisant une estimation par les moindres carrés.
- Procédé selon la revendication 10, comprenant en outre la normalisation de la corrélation croisée.
- Procédé selon la revendication 11 ou 12, comprenant en outre une post-mise à l'échelle (514) du signal de composante de parole estimé et du signal de composante d'ambiance estimé.
- Procédé selon l'une quelconque des revendications 10 à 13, comprenant en outre :la synthèse (516) des signaux en sous-bandes en utilisant les deuxièmes puissances estimées et un gain spécifié par l'utilisateur, dans lequel le gain inclut le gain spécifié par l'utilisateur et la génération du signal audio modifié comprend la conversion (518) des signaux en sous-bandes synthétisés en un signal audio dans le domaine temporel ayant un signal de composante de parole qui est modifié par le gain spécifié par l'utilisateur.
- Appareil de traitement d'un signal audio, comprenant:une interface (602) configurable pour obtenir un signal audio à canaux multiples incluant un signal de composante de parole et un signal d'une autre composante ;une interface de saisie d'utilisateur (610) configurable pour recevoir une information se rapportant à un gain pour commander un niveau du signal de composante de parole ;un estimateur de puissances (622) configurable pour déterminer des valeurs de gain pour au moins deux canaux du signal audio à canaux multiples, les valeurs de gain représentant un niveau pour chaque canal des au moins deux canaux;un estimateur de signaux (624) configurable pour :déterminer une corrélation croisée entre les au moins deux canaux,déterminer une localisation spatiale du signal de composante de parole en utilisant au moins l'une de la corrélation croisée et des valeurs de gain, etidentifier le signal de composante de parole sur la base de la localisation spatiale du signal de composante de parole ;un synthétiseur de signaux (628) couplé à l'estimateur de signaux et configurable pour :modifier le signal de composante de parole en appliquant une valeur de gain au signal de composante de parole, etgénérer un signal audio modifié incluant le signal de composante de parole modifié ; etune unité de sortie (608) configurable pour délivrer en sortie le signal audio modifié.
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PCT/EP2007/008028 WO2008031611A1 (fr) | 2006-09-14 | 2007-09-14 | Techniques facilitant le dialogue |
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EP2070389A1 EP2070389A1 (fr) | 2009-06-17 |
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EP07858967A Not-in-force EP2070391B1 (fr) | 2006-09-14 | 2007-09-14 | Techniques de renforcement des dialogues |
EP07802317A Not-in-force EP2070389B1 (fr) | 2006-09-14 | 2007-09-14 | Techniques facilitant le dialogue |
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EP07858967A Not-in-force EP2070391B1 (fr) | 2006-09-14 | 2007-09-14 | Techniques de renforcement des dialogues |
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BR (1) | BRPI0716521A2 (fr) |
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DE (1) | DE602007010330D1 (fr) |
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- 2007-09-14 US US11/855,570 patent/US8184834B2/en not_active Expired - Fee Related
- 2007-09-14 MX MX2009002779A patent/MX2009002779A/es not_active Application Discontinuation
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